similar to: DONT_OPTIMISE, BETTER_BACKTRACES and performance

I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so often asterisk crashes and then restarts. I am not seeing any core dumps on the box. The only I thing I see every time is a second before Asterisk crashes there is a AAAA lookup for the boxes hostname. As soon as it gets the response I see that asterisk is restarting. Any idea what would cause this and how would get a dump or

I have been following the "Asterisk CLI unresponsive" thread and saw the suggestion to enable BETTER_BACKTRACES when I went in to do that it said it requires libbfd.. When I went to add that from yast I get no packages by that name. I am on openSUSE an ideas if this is part of another package or how I would get it? Thanks Bryant -------------- next part -------------- An HTML

Hello Asterisk list, I am facing some Asterisk crashes which are consistently pointing to the same backtrace, which is the following (using DONT_OPTIMIZE, BETTER_BACKTRACES and MALLOC_DEBUG): Thread 1 (Thread 0x7f1f08be8700 (LWP 1767)): #0 0x00007f1f9bed3395 in __strcasecmp_l_sse42 () from /lib64/libc.so.6 #1 0x00000000004a91ca in cdr_object_get_by_name_cb () #2 0x0000000000463c60 in

Hello, I've got a 13.13.1 system using PJSIP stack on debian Jessie. It runs from 50 to 100 simultaneous calls (so 100 to 200 PJSIP channels) all day long. >From time to time, roughly meaning once a month, it segfaults with lines (from dmesg -T output) like this: asterisk[1160]: segfault at 7efffffe ip 00000000005881d6 sp 00007fec95c33910 error 4 in asterisk[400000+2a2000] Debug level

On Tue, Jun 26, 2018 at 7:59 PM, Richard Mudgett <rmudgett at digium.com> wrote: > > > On Tue, Jun 26, 2018 at 6:15 PM, Dovid Bender <dovid at telecurve.com> wrote: > >> I have Asterisk running on a Ubuntu 18.0.4 on Digital Ocean. Every so >> often asterisk crashes and then restarts. I am not seeing any core dumps on >> the box. The only I thing I see

Hello asterisk-users, We've been having intermittent issues with chan_sip - it stops responding to cli requests, trying to reload chan_sip from cli doesn't seem to have any effect, initiated calls carry on for a short period, but no new SIP requests are processed ('sip show channels' hangs forever, server stops responding to SIP OPTIONS, or any other SIP messages). We have updated

Thanks for you answer Marcus, So maybe this means some bug was fixed? Anyone aware of something related? >From the release notes, I couldn't find any direct change that could fix this.... Thanks, Kind regards, Patrick Wakano On 21 February 2018 at 20:29, Marcus Kvarsell <Marcus.Kvarsell at fogwise.se> wrote: > Hello, i found upgrading to asterisk 15 helped. > > > >

I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a "core restart now" cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2

Hello list, Hope you are all doing well! I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and I wonder if someone can put some light on it. Log history short, install_prereq fails to install the packages (not sure how important they actually are....): speexdsp-devel, gmime-devel, uriparser-devel, iksemel-devel, uw-imap-devel, hoard Then, I am running the following commands

2016-09-06 17:48 GMT+02:00 Tzafrir Cohen <tzafrir.cohen at xorcom.com>: > On Tue, Sep 06, 2016 at 06:37:52AM -0600, George Joseph wrote: > > On Tue, Sep 6, 2016 at 1:55 AM, Olivier <oza.4h07 at gmail.com> wrote: > > > > Where should core file be created when Asterisk is run as a daemon by > > > asterisk user and group ? > > > Is there a setting I

On Tue, Feb 14, 2017 at 2:51 PM, George Joseph <gjoseph at digium.com> wrote: > > > On Tue, Feb 14, 2017 at 10:21 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> I've got a 13.13.1 system using PJSIP stack on debian Jessie. >> It runs from 50 to 100 simultaneous calls (so 100 to 200 PJSIP channels) >> all day long. >>

On Fri, Feb 17, 2017 at 5:17 AM, Olivier <oza.4h07 at gmail.com> wrote: > Hi George, > > How does ast_coredumper compare to ast_grab_core ) ? > Is it worth learning to use both or shall favor one ? > > PS: As I don't know either program, yet, my question may seem silly. > Please, forgive me for this > Not silly at all. ast_grab_core actually kills asterisk to

I think were getting closer: I did: - I edited /etc/default/asterisk to include : AST_USER="root" AST_GROUP="root" # systemctl daemon-reload # systemctl start asterisk # ps aux | grep asterisk root 3602 7.1 2.5 60332 26012 ? Ssl 16:00 0:03 /usr/sbin/asterisk -U root -G root -g # rasterisk # pkill -SEGV asterisk Then console showed: Segmentation error (core

Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn?t a "typo? error of timers parameters, i have an error on global tag and can?t load the timers I was getting this : [Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_text_line: parse error: No category context for line 631 of /etc/asterisk/pjsip.conf after fix global issue

We have a client that has sporadic problems with the *8 pickup facility. The server they are using is 1.8.5 and they are using Snom phones. Every now and then when they try to do a pickup from another phone they get a forbidden message on the phone and I can see the following in the logs. [Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL [Aug 8 11:51:53] ERROR[19314] astobj2.c:

Hi all, When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately. At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc. Thanks in advance, Giang -------------- next part -------------- An HTML

Hi Is there a way I can stop Manager logged on/off messages from going to the console/logs without losing all the other information I need? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Hi I'm using asterisk 1.8 Does anyone know how to use the queue reload command. The built in help doesn't really help. queue reload {parameters|membe Reload queues, members, queue rules, or parameters Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk

It there a native asterisk dialplan function which will tell me the position of a specific character in a given string? eg if I wanted to find what position the '@' was at in ${SIPURI} Thanks in advance Ish -- Ishfaq Malik <ish at pack-net.co.uk> Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w:

Hi I have some question about mysql realtime. 1) Anyone know exactly if there is a specific order to declare sip table column for realtime ? In which file can I find that order ? 2) In my extconfig.conf, [settings] are : sipusers => mysql,general,siptable sippeers => mysql,general,siptable so means that I use realtime dynamic exactly ? Is it normal if some parameters from sip.conf still